[asterisk-dev] asterisk-dev Digest, Vol 36, Issue 69
jazzy singh
jazzy_gill at yahoo.com
Thu Jul 26 18:26:52 CDT 2007
I've been doing everything the forum member's been suggesting me, I even attached my conf files as well. But I haven't gotten anywhere that's the only reason I'm keep on replying to these emails...sorry
asterisk-dev-request at lists.digium.com wrote: Send asterisk-dev mailing list submissions to
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When replying, please edit your Subject line so it is more specific
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Today's Topics:
1. Re: plese help (Russell Bryant)
2. Re: plese help (Matthew Rubenstein)
3. (Update) Digium FTP server will be replaced with HTTP server
(Kevin P. Fleming)
4. Re: asterisk-dev Digest, Vol 36, Issue 68 (jazzy singh)
5. Re: asterisk broke and I'm getting fired from
(Eric "ManxPower" Wieling)
6. Re: asterisk-dev Digest, Vol 36, Issue 68 (Russell Bryant)
----------------------------------------------------------------------
Message: 1
Date: Thu, 26 Jul 2007 17:51:43 -0400
From: Russell Bryant
Subject: Re: [asterisk-dev] plese help
To: Asterisk Developers Mailing List
Message-ID: <46A9176F.1090400 at digium.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
jazzy singh wrote:
> Please help......I'm using asterisk-1.2.13 with lumenvox, festival.. I
> was able to compile and install it without any problem,,as soon as I run
> asterisk it dies without giving any kind of error. Anybody please help ....
You have got to be kidding me ... You have now started 4 threads asking for
help on the same thing. You have received many suggestions for things to check
for, and you have not responded to most of them.
This is not intended to be a technical support forum. Please try the
asterisk-users mailing list to get help. If you are able to determine a problem
*with an up to date version*, then you can file a bug on bugs.digium.com. We
don't handle bug reports through this mailing list so that the information stays
organized and doesn't get lost.
--
Russell Bryant
Software Engineer
Digium, Inc.
------------------------------
Message: 2
Date: Thu, 26 Jul 2007 17:55:28 -0400
From: Matthew Rubenstein
Subject: Re: [asterisk-dev] plese help
To: David Boyd
Cc: Asterisk -Dev
Message-ID: <1185486928.21464.153.camel at pont>
Content-Type: text/plain
If he can't even ask for Asterisk help properly, including find the
right support list (-users), even after told to do so, then maybe he
should get fired for leaving his boss' phone system down for over a
whole business day. Not everyone is cut out to do this themselves - some
people need to hire consultants to help them, or at least to train them,
or to do both. Like how to plan for inevitable outages.
On Thu, 2007-07-26 at 16:49 -0500, asterisk-dev-request at lists.digium.com
wrote:
> Date: Thu, 26 Jul 2007 17:44:51 -0400
> From: David Boyd
> Subject: Re: [asterisk-dev] plese help
> To: Asterisk Developers Mailing List
> Message-ID: <1185486292.9441.45.camel at d9100>
> Content-Type: text/plain
>
> Hi Jazzy,
>
> Why don't you help yourself and post configs, versions, changes that
> you
> made to the system, copies of error and logs etc. and someone might
> help
> you. However I am of the opinion that you don't really want assistance
> you simply are having fun screwing with everybody else.
>
> Next point the application can't really die without any errors at
> all.
> There has to be some indication of what took place, so I go back to my
> original statement that you are just screwing with mine and everybody
> else' time.
>
>
>
> Dave
--
(C) Matthew Rubenstein
------------------------------
Message: 3
Date: Thu, 26 Jul 2007 17:17:29 -0500
From: "Kevin P. Fleming"
Subject: [asterisk-dev] (Update) Digium FTP server will be replaced
with HTTP server
To: Asterisk Developers Mailing List
Message-ID: <46A91D79.7000807 at digium.com>
Content-Type: text/plain; charset=ISO-8859-1
Some time in the next two weeks, Digium will be shutting down our FTP
server, located at ftp.digium.com, and begin using only the existing
HTTP server on the same system instead.
We have decided to only offer our public downloads over the HTTP
protocol, not the FTP protocol, primarily for reasons related to our
marketing department :-)
The site has a new name, downloads.digium.com, and will also respond to
the old name ftp.digium.com, but will no longer respond to requests made
via the FTP protocol; only the HTTP protocol will be supported. There
should be no other user-visible changes when this change is made to the
server. We will begin using the downloads.digium.com name in all our
announcements, security advisories, documentation and other places
immediately, and encourage the community to do so as well.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
------------------------------
Message: 4
Date: Thu, 26 Jul 2007 15:32:49 -0700 (PDT)
From: jazzy singh
Subject: Re: [asterisk-dev] asterisk-dev Digest, Vol 36, Issue 68
To: asterisk-dev at lists.digium.com
Message-ID: <465433.11341.qm at web50703.mail.re2.yahoo.com>
Content-Type: text/plain; charset="iso-8859-1"
Hey David,
I apologize if you thought that I'm wasting everyone's time overhere ...I'm not...I'm under a lot of frustration right now....I just want this thing to start working like it was till yesterday. I'm a newbie at asterisk so I'm discovering things as I go I figure out how to make it print error messages when i restart here'we go
version = I'm using asterisk-1.2.13 (it worked fine till last nite..when I was testing it and must've called 20, 30)
my version of zaptel -1.2.10
lumenvox7.5
this is the error I get when starting asterisk
Asterisk Dynamic Loader Starting:
== Parsing '/etc/asterisk/modules.conf': Found
[res_musiconhold.so] => (Music On Hold Resource)
== Registered application 'MusicOnHold'
== Registered application 'WaitMusicOnHold'
== Registered application 'SetMusicOnHold'
== Registered application 'StartMusicOnHold'
== Registered application 'StopMusicOnHold'
== Parsing '/etc/asterisk/musiconhold.conf': Found
[res_speech.so] => (Lumenvox Speech Recognition)
Jul 26 14:39:09 NOTICE[5770]: res_speech_lumenvox.c:758 load_module: Lumenvox SRE Connector module Copyright (C) 1999-2007 Digium, Inc.
Jul 26 14:39:09 NOTICE[5770]: res_speech_lumenvox.c:759 load_module: This module is supplied under a commercial license granted by Digium, Inc.
== Parsing '/etc/asterisk/lumenvox.conf': Found
-- Using server(s): 127.0.0.1
-- Loaded grammar 'plan'
-- Loaded grammar 'one_to_ten'
-- Loaded tweaking profile default
asterisk: symbol lookup error: /usr/lib/asterisk/modules/res_speech.so: undefined symbol: ast_speech_register
this is what my modules.conf looks like...
;
; Asterisk configuration file
;
; Module Loader configuration file
;
[modules]
autoload=yes
;
; Any modules that need to be loaded before the Asterisk core has been
; initialized (just after the logger has been initialized) can be loaded
; using 'preload'. This will frequently be needed if you wish to map all
; module configuration files into Realtime storage, since the Realtime
; driver will need to be loaded before the modules using those configuration
; files are initialized.
;
; An example of loading ODBC support would be:
;preload => res_odbc.so
;preload => res_config_odbc.so
;
; If you want, load the GTK console right away.
; Don't load the KDE console since
; it's not as sophisticated right now.
;
noload => pbx_gtkconsole.so
;load => pbx_gtkconsole.so
noload => pbx_kdeconsole.so
;
; Intercom application is obsoleted by
; chan_oss. Don't load it.
;
noload => app_intercom.so
;
; The 'modem' channel driver and its subdrivers are
; obsolete, don't load them.
;
noload => chan_modem.so
noload => chan_modem_aopen.so
noload => chan_modem_bestdata.so
noload => chan_modem_i4l.so
;
load => res_musiconhold.so
;
; Load either OSS or ALSA, not both
; By default, load OSS only (automatically) and do not load ALSA
;
noload => chan_alsa.so
load => res_speech.so
load => res_speech_lumenvox.so
;noload => chan_oss.so
;
; Module names listed in "global" section will have symbols globally
; exported to modules loaded after them.
;
[global]
this is what my lumenvox.com looks like
; Lumenvox configuration file
[general]
servers=127.0.0.1 ; SRE Servers to use
save_sound_files=no ; Set to yes to save sound files for tuner usage
; Pre-loaded grammars
[grammars]
plan=/etc/asterisk/grammars/plan.gram
one_to_ten=/etc/asterisk/grammars/one_to_ten.gram
; Lumenvox tweaking profiles
; A tweaking profile can be used by using the SPEECH_ENGINE dialplan function. For example,
; to apply the default profile you would use Set(SPEECH_ENGINE(profile)=default)
; NOTE: Each option can also be set in the dialplan by using Set(SPEECH_ENGINE(name)=value)
; Example: Set(SPEECH_ENGINE(vad_eos_delay)=100)
; Default settings
[default]
; SNR threshold/Barge-In Level.
; An audio frame will be considered for voice activity only when the SNR metric is higher than this threshold.
; So for noisy channel, this value should be lower, so that it is easier to barge in.
; NOTE: this value is not a measurement in dB. It is just a relative value compared to an internal standard.
vad_bargein_level=40
; End-of-speech delay in ms.
; The initial audio used to initialize voice activity detector.
; Setting this parameter to 60 means using the first 60ms as initialization data.
vad_eos_delay=2000
; Noise floor threshold.
; An audio frame will be considered for voice activity only when the average energy
; is higher than this threshold. The default value is 200. This parameter is
; particularly useful when the echo canceler doesn't work very well.
; Because the residual echo has all features of voice, so the only way we
; can filter them out is using a hard constraint on energy.
vad_noise_floor=200
; Wind back.
; The length of audio wound back from the point of voice detection.
; It helps in the situation of weak speech onset. The resolution of this
; parameter is 1/8 sec,i.e. 125ms, which means setting this value to 249ms
; is same as setting it to 125ms.
vad_wind_back=255
; Burst control threshold in ms.
; Barge-in will be triggered only when the duration of voice is longer
; than this threshold.
vad_burst_threshold=100
; After barge-in, the streaming interface will flag that speech was detected
; if it detected it in the time frame specified by this option.
end_of_speech_timeout=8000
; Whether to use the out-of-vacabulary filter during decode.
use_oov_filter=no
---------------------------------
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------------------------------
Message: 5
Date: Thu, 26 Jul 2007 17:55:05 -0500
From: "Eric \"ManxPower\" Wieling"
Subject: Re: [asterisk-dev] asterisk broke and I'm getting fired from
To: Asterisk Developers Mailing List
Message-ID: <46A92649.1080305 at fnords.org>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Start Asterisk as "asterisk -cvvv"
jazzy singh wrote:
> I did that but it still is dying,,,,just to let you guys know I'm using lumenvox as well, when I restart asterisk it restarts and then it just dies right away without any error message. Please help
> Thanks
>
>
>
>
> Date: Thu, 26 Jul 2007 14:20:59 -0400
> From: "harish kasiviswanathan"
> Subject: Re: [asterisk-dev] asterisk broke and I'm getting fired from
> my job :(....
> To: "Asterisk Developers Mailing List"
> Message-ID:
> <8030b59a0707261120u61afd7dax72aa99be40c3e9f0 at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-13"
>
> Looks like you are using the old version of zaptel driver and new version of
> asterisk. Upgrade to zaptel-1.4.4 or newer and this error should go away.
>
> harish
>
> On 7/26/07, jazzy singh wrote:
>> I was running a successfull asterisk box but i don't know what happened
>> for some reason asterisk shuts down as soon as I start it. It won't give me
>> any error or anything, but last nite when I was forking through error log I
>> did see this though
>> Jul 25 23:02:55 NOTICE[8329] chan_zap.c: Got event 18 (Ring Begin)...
>> Jul 25 23:02:55 WARNING[6651] channel.c: Avoided initial deadlock for
>> '0x8a94f88', 10 retries!
>> Jul 25 23:17:39 NOTICE[9303] chan_zap.c: Got event 18 (Ring Begin)...
>>
>> and then i tried to recompile asterisk and it gave me this
>>
>> codec_zap.c: In function ?find_transcoders?:
>> codec_zap.c:833: error: variable ?info? has initializer but incomplete
>> type
>> codec_zap.c:833: warning: excess elements in struct initializer
>> codec_zap.c:833: warning: (near initialization for ?info?)
>> codec_zap.c:833: error: storage size of ?info? isn?t known
>> codec_zap.c:838: error: ?ZT_TCOP_GETINFO? undeclared (first use in this
>> function
>> )
>> codec_zap.c:848: error: ?ZT_TRANSCODE_OP? undeclared (first use in this
>> function
>> )
>> codec_zap.c:833: warning: unused variable ?info?
>> make[1]: *** [codec_zap.o] Error 1
>> make[1]: Leaving directory `/downloads/asterisk-1.2.20/codecs'
>> make: *** [subdirs] Error 1
>>
>>
>> I was able to compile and install the same source code before. Please
>> someone help me if I don't get this to work today I might get fired :(..
>> please help.....
>> thanks in advance
>>
>>
>> ------------------------------
>
>
>
>
>
>
>
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------------------------------
Message: 6
Date: Thu, 26 Jul 2007 18:55:45 -0400
From: Russell Bryant
Subject: Re: [asterisk-dev] asterisk-dev Digest, Vol 36, Issue 68
To: Asterisk Developers Mailing List
Message-ID: <46A92671.2000605 at digium.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
jazzy singh wrote:
> I apologize if you thought that I'm wasting everyone's time
> overhere ...I'm not...I'm under a lot of frustration right now....I just
> want this thing to start working like it was till yesterday. I'm a
> newbie at asterisk so I'm discovering things as I go I figure out how to
> make it print error messages when i restart here'we go
I already asked you once. Please move to the asterisk-users mailing list.
People could easily help you there. This list is for development discussion.
> version = I'm using asterisk-1.2.13 (it worked fine till last nite..when
> I was testing it and must've called 20, 30)
> my version of zaptel -1.2.10
> lumenvox7.5
I also already noted that you need to update to a version that isn't 9 months
old before reporting that there is a problem.
--
Russell Bryant
Software Engineer
Digium, Inc.
------------------------------
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End of asterisk-dev Digest, Vol 36, Issue 69
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